There is a growing need for automatic recognition of music or other audio signals generated from a variety of sources. For example, owners of copyrighted works or advertisers are interested in obtaining data on the frequency of broadcast of their material. Music tracking services provide playlists of major radio stations in large markets. Consumers would like to identify songs or advertising broadcast on the radio, so that they can purchase new and interesting music or other products and services. Any sort of continual or on-demand sound recognition is inefficient and labor intensive when performed by humans. An automated method of recognizing music or sound would thus provide significant benefit to consumers, artists, and a variety of industries. As the music distribution paradigm shifts from store purchases to downloading via the Internet, it is quite feasible to link directly computer-implemented music recognition with Internet purchasing and other Internet-based services.
Traditionally, recognition of songs played on the radio has been performed by matching radio stations and times at which songs were played with playlists provided either by the radio stations or from third party sources. This method is inherently limited to only radio stations for which information is available. Other methods rely on embedding inaudible codes within broadcast signals. The embedded signals are decoded at the receiver to extract identifying information about the broadcast signal. The disadvantage of this method is that special decoding devices are required to identify signals, and only those songs with embedded codes can be identified.
Any large-scale audio recognition requires some sort of content-based audio retrieval, in which an unidentified broadcast signal is compared with a database of known signals to identify similar or identical database signals. Note that content-based audio retrieval is different from existing audio retrieval by web search engines, in which only the metadata text surrounding or associated with audio files is searched. Also note that while speech recognition is useful for converting voiced signals into text that can then be indexed and searched using well-known techniques, it is not applicable to the large majority of audio signals that contain music and sounds. In some ways, audio information retrieval is analogous to text-based information retrieval provided by search engines. In other ways, however, audio recognition is not analogous: audio signals lack easily identifiable entities such as words that provide identifiers for searching and indexing. As such, current audio retrieval schemes index audio signals by computed perceptual characteristics that represent various qualities or features of the signal.
Content-based audio retrieval is typically performed by analyzing a query signal to obtain a number of representative characteristics, and then applying a similarity measure to the derived characteristics to locate database files that are most similar to the query signal. The similarity of received objects is necessarily a reflection of the perceptual characteristics selected. A number of content-based retrieval methods are available in the art. For example, U.S. Pat. No. 5,210,820, issued to Kenyon, discloses a signal recognition method in which received signals are processed and sampled to obtain signal values at each sampling point. Statistical moments of the sampled values are then computed to generate a feature vector that can be compared with identifiers of stored signals to retrieve similar signals. U.S. Pat. Nos. 4,450,531 and 4,843,562, both issued to Kenyon et al., disclose similar broadcast information classification methods in which cross-correlations are computed between unidentified signals and stored reference signals.
A system for retrieving audio documents by acoustic similarity is disclosed in J. T. Foote, “Content-Based Retrieval of Music and Audio,” in C.-C. J. Kuo et al., editor, Multimedia Storage and Archiving Systems II, Proc. of SPIE, volume 3229, pages 138 147, 1997.Feature vectors are calculated by parameterizing each audio file into mel-scaled cepstral coefficients, and a quantization tree is grown from the parameterization data. To perform a query, an unknown signal is parameterized to obtain feature vectors that are then sorted into leaf nodes of the tree. A histogram is collected for each leaf node, thereby generating an N-dimensional vector representing the unknown signal. The distance between two such vectors is indicative of the similarity between two sound files. In this method, the supervised quantization scheme learns distinguishing audio features, while ignoring unimportant variations, based on classes into which the training data are assigned by a human. Depending upon the classification system, different acoustic features are chosen to be important. Thus this method is more suited for finding similarities between songs and sorting music into classes than it is to recognizing music.
A method for content-based analysis, storage, retrieval, and segmentation of audio information is disclosed in U.S. Pat. No. 5,918,223, issued to Blum et al. In this method, a number of acoustical features, such as loudness, bass, pitch, brightness, bandwidth, and Mel-frequency cepstral coefficients, are measured at periodic intervals of each file. Statistical measurements of the features are taken and combined to form a feature vector. Audio data files within a database are retrieved based on the similarity of their feature vectors to the feature vector of an unidentified file.
A key problem of all of the above prior art audio recognition methods is that they tend to fail when the signals to be recognized are subject to linear and nonlinear distortion caused by, for example, background noise, transmission errors and dropouts, interference, band-limited filtering, quantization, time-warping, and voice-quality digital compression. In prior art methods, when a distorted sound sample is processed to obtain acoustical features, only a fraction of the features derived for the original recording are found. The resulting feature vector is therefore not very similar to the feature vector of the original recording, and it is unlikely that correct recognition can be performed. There remains a need for a sound recognition system that performs well under conditions of high noise and distortion.
Another problem with prior art methods is that they are computationally intensive and do not scale well. Real-time recognition is thus not possible using prior art methods with large databases. In such systems, it is unfeasible to have a database of more than a few hundred or thousand recordings. Search time in prior art methods tends to grow linearly with the size of the database, making scaling to millions of sounds recordings economically unfeasible. The methods of Kenyon also require large banks of specialized digital signal processing hardware.
Existing commercial methods often have strict requirements for the input sample to be able to perform recognition. For example, they require the entire song or at least 30 seconds of the song to be sampled or require the song to be sampled from the beginning. They also have difficulty recognizing multiple songs mixed together in a single stream. All of these disadvantages make prior art methods unfeasible for use in many practical applications.